florian-h05 / webrtc-sip-gwLinks
A WebRTC-SIP gateway for AVM Fritz!Box based on Kamailio and rtpengine.
☆23Updated 2 months ago
Alternatives and similar repositories for webrtc-sip-gw
Users that are interested in webrtc-sip-gw are comparing it to the libraries listed below
Sorting:
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- Connect as WebRTC with FreeSWITCH using SIPjs☆11Updated 6 years ago
- Black Box SIP Tester☆31Updated last year
- ☆16Updated 9 years ago
- baresip python wrapper☆60Updated 10 months ago
- Simple SIP command line Softphone Client☆60Updated 7 years ago
- Skype for Business/Lync PSTN gateway☆24Updated 6 years ago
- Asterisk Application - Two-way audio with your Camera using RTSP and SIP to an Asterisk Channel☆31Updated last week
- SIP Phone WebRTC for your browser☆59Updated 4 years ago
- ☆45Updated last year
- Telegram <-> SIP voice gateway☆24Updated last year
- React SIP user agent☆56Updated 6 years ago
- Dockerfile for creating a minimal Freeswitch image for use with drachtio-mrf☆20Updated last year
- ☆16Updated 3 years ago
- SIP Phone Learning Tool☆43Updated 4 years ago
- Classes of Telephony for the Python's diagrams package☆11Updated 4 years ago
- WebRTC SIP based VoIP client software (+chrome extension)☆111Updated last year
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆37Updated 7 years ago
- ☆90Updated last week
- SIP outbound proxy based on drachtio and freeswitch that includes siprec client functionality☆20Updated last year
- Realtime configuration engine and CDR&CEL backend for Asterisk with MongoDB☆36Updated 7 years ago
- OpenSIPS + RTPEngine Recording + Speech Recognition in HEP☆21Updated last year
- Kamailio config for public/private proxy with rtpengine☆15Updated 2 years ago
- ☆36Updated last month
- jambones REST APIs☆24Updated last week
- Dockerfile for creating a minimal Freeswitch image for use with drachtio-mrf☆10Updated last year
- A curated list of HEP / EEP enabled projects☆28Updated 6 years ago
- SIP provisioning server / Auto configuration system (ACS)☆30Updated 3 years ago
- Call API is a front-end layer for managing advanced SIP call flows. It listens for WebSocket connections and talks JSON-RPC 2.0 over them…☆56Updated last month
- Asterisk to MQTT Bridge☆34Updated 11 years ago